libbkmz/dongle.comf Secret

## dongle.comf
[general]

interval=10                     ; Number of seconds between trying to connect to devices

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes                 ; Enables the use of a jitterbuffer on the receiving side of a
                                ; Dongle channel. Defaults to "no". An enabled jitterbuffer will
                                ; be used only if the sending side can create and the receiving
                                ; side can not accept jitter. The Dongle channel can't accept jitter,
                                ; thus an enabled jitterbuffer on the receive Dongle side will always
                                ; be used if the sending side can create jitter.

;jbforce = no                   ; Forces the use of a jitterbuffer on the receive side of a Dongle
                                ; channel. Defaults to "no".

;jbmaxsize = 200                ; Max length of the jitterbuffer in milliseconds.

;jbresyncthreshold = 1000       ; Jump in the frame timestamps over which the jitterbuffer is
                                ; resynchronized. Useful to improve the quality of the voice, with
                                ; big jumps in/broken timestamps, usually sent from exotic devices
                                ; and programs. Defaults to 1000.

;jbimpl = fixed                 ; Jitterbuffer implementation, used on the receiving side of a Dongle
                                ; channel. Two implementations are currently available - "fixed"
                                ; (with size always equals to jbmaxsize) and "adaptive" (with
                                ; variable size, actually the new jb of IAX2). Defaults to fixed.

;jbtargetextra = 40             ; This option only affects the jb when 'jbimpl = adaptive' is set.
                                ; The option represents the number of milliseconds by which the new jitter buffer
                                ; will pad its size. the default is 40, so without modification, the new
                                ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
                                ; increasing this value may help if your network normally has low jitter,
                                ; but occasionally has spikes.

;jblog = no                     ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

[defaults]
; now you can set here any not required device settings as template
;   sure you can overwrite in any [device] section this default values

context=default                 ; context for incoming calls
group=0                         ; calling group
rxgain=3                        ; increase the incoming volume; may be negative
txgain=3                        ; increase the outgoint volume; may be negative
autodeletesms=yes               ; auto delete incoming sms
resetdongle=yes                 ; reset dongle during initialization with ATZ command
u2diag=-1                       ; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres=yes              ; use the caller ID presentation or not
callingpres=allowed_passed_screen ; set caller ID presentation          by default use default network settings
disablesms=no                   ; disable of SMS reading from device when received
                                ;  chan_dongle has currently a bug with SMS reception. When a SMS gets in during a
                                ;  call chan_dongle might crash. Enable this option to disable sms reception.
                                ;  default = no

language=en                     ; set channel default language
smsaspdu=yes                    ; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
mindtmfgap=45                   ; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration=80              ; minimal DTMF tone duration in ms
mindtmfinterval=200             ; minimal interval between ends of DTMF of same digits in ms

callwaiting=auto                ; if 'yes' allow incoming calls waiting; by default use network settings
                                ; if 'no' waiting calls just ignored
disable=no                      ; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section

initstate=start                 ; specified initial state of device, must be one of 'stop' 'start' 'remote'
                                ;   'remove' same as 'disable=yes'

exten=+375441111111             ; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)

dtmf=inband                     ; control of incoming DTMF detection, possible values:
                                ;   off    - off DTMF tones detection, voice data passed to asterisk unaltered
                                ;              use this value for gateways or if not use DTMF for AVR or inside dialplan
                                ;   inband - do DTMF tones detection
                                ;   relax  - like inband but with relaxdtmf option
                                ;  default is 'relax' by compatibility reason

; dongle required settings
[dongle0]
audio=/dev/ttyUSB1              ; tty port for audio connection;        no default value
data=/dev/ttyUSB2               ; tty port for AT commands;             no default value

context=dongle-incoming; контекст
group=0

; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345
;  imei and imsi must contain exactly 15 digits !
;  imei/imsi discovery is available on Linux only
imei=123456789012345 ; выставлены - правильно
imsi=123456789012345 ; выставлены - правильно - здесь не показываю просто

; if audio and data set together with imei and/or imsi audio and data has precedence
;   you can use both imei and imsi together in this case exact match by imei and imsi required

## extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp


[dongle-incoming-ussd]
exten => ussd,1,Noop(Incoming USSD: ${BASE64_DECODE(${USSD_BASE64})})
exten => ussd,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME}: ${BASE64_DECODE(${USSD_BASE64})}' >> /var/log/asterisk/ussd.txt)
exten => ussd,n,Hangup()

[dongle-incoming-sms]
exten => sms,1,Noop(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}' >> /var/log/asterisk/sms.txt)
exten => sms,n,Hangup()

[dongle-incoming]
include => dongle-incoming-ussd
include => dongle-incoming-sms

;exten => +375441111111,1,Hangup()

exten => +375441111111,1,NoOp()
same => n,Wait(1)
;same => n,Answer()
;same => n,WaitExten(35)


;exten => _7XXXXXXXXXX,1,NoOp()
;same => n,Dial(SIP/10)
;same => n,Dial(SIP/74951234567@zadarma)
same => n,Dial(SIP/00000@zadarma)

same => n,Hangup()


[default]
exten => 10,1,Dial(Dongle/dongle0/1111111)
exten => 00000,1,Dial(SIP/00000@zadarma)

## sip.conf
[general]

limitonpeer=yes
notifyringing=yes
notifyhold=yes
allowsubscribe=yes


context=default
allowguest=no           ; no guests
allowoverlap=no         ; allow call with additional numbers -> no
alwaysauthreject=yes    ; all errors from client gives to him as auth errors(yes), no - disabling this feature
; defaultexpiry=360     ; interval mandatory registration
callevents=yes          ; for future. For activating music on hold
tcpenable=yes           ; enabling TCP for SIP
rtptimeout=60           ; timeout before kick
language=ru             ; language for voice subsystem
bindport=5060
bindaddr=0.0.0.0
;srvlookup=yes          ; searching for server address by username <testuser@domain.com>
tos_sip=cs3
tos_audio=ef

match_auth_username=yes

;video=yes
;videosupport=yes

; echocancel=yes
; echocancelwhenbridged=yes

disallow=all
;allow=alaw
;allow=g729
;allow=g723
;allow=ulaw

canreinvite=no


pedantic=no

[zadarma-tpl](!)
type=friend
host=sipde.zadarma.com
fromdomain=sip.zadarma.com
nat=force_rport,comedia
;directmedia=nonat
;nat=yes
dtmfmode=rfc2833
insecure=invite
canreinvite=no
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm

[zadarma](zadarma-tpl)
context=from-zadarma
username=11111
secret=xxxxx
fromuser=11111


[defaults](!)
 canreinvite=no
 disallow=all
 ;allow=g729
 allow=alaw
 allow=ulaw
 ;allow=ilbc
 ;allow=g726
 ;allow=g723
 ;allow=g722
 allow=gsm
nat=force_rport,comedia
directmedia=nonat

[peer](!,defaults)
 type=friend
 dtmfmode=rfc2833
 host=dynamic
 qualify=yes
 call-limit=3
 busylevel=1
 rtpkeepalive=5


[10](peer) ; username
 callerid=username
 defaultuser=username
 secret=00000
	[general]

	interval=10 ; Number of seconds between trying to connect to devices

	;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
	;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
	; Dongle channel. Defaults to "no". An enabled jitterbuffer will
	; be used only if the sending side can create and the receiving
	; side can not accept jitter. The Dongle channel can't accept jitter,
	; thus an enabled jitterbuffer on the receive Dongle side will always
	; be used if the sending side can create jitter.

	;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a Dongle
	; channel. Defaults to "no".

	;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

	;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
	; resynchronized. Useful to improve the quality of the voice, with
	; big jumps in/broken timestamps, usually sent from exotic devices
	; and programs. Defaults to 1000.

	;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a Dongle
	; channel. Two implementations are currently available - "fixed"
	; (with size always equals to jbmaxsize) and "adaptive" (with
	; variable size, actually the new jb of IAX2). Defaults to fixed.

	;jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
	; The option represents the number of milliseconds by which the new jitter buffer
	; will pad its size. the default is 40, so without modification, the new
	; jitter buffer will set its size to the jitter value plus 40 milliseconds.
	; increasing this value may help if your network normally has low jitter,
	; but occasionally has spikes.

	;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
	;-----------------------------------------------------------------------------------

	[defaults]
	; now you can set here any not required device settings as template
	; sure you can overwrite in any [device] section this default values

	context=default ; context for incoming calls
	group=0 ; calling group
	rxgain=3 ; increase the incoming volume; may be negative
	txgain=3 ; increase the outgoint volume; may be negative
	autodeletesms=yes ; auto delete incoming sms
	resetdongle=yes ; reset dongle during initialization with ATZ command
	u2diag=-1 ; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
	usecallingpres=yes ; use the caller ID presentation or not
	callingpres=allowed_passed_screen ; set caller ID presentation by default use default network settings
	disablesms=no ; disable of SMS reading from device when received
	; chan_dongle has currently a bug with SMS reception. When a SMS gets in during a
	; call chan_dongle might crash. Enable this option to disable sms reception.
	; default = no

	language=en ; set channel default language
	smsaspdu=yes ; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
	mindtmfgap=45 ; minimal interval from end of previews DTMF from begining of next in ms
	mindtmfduration=80 ; minimal DTMF tone duration in ms
	mindtmfinterval=200 ; minimal interval between ends of DTMF of same digits in ms

	callwaiting=auto ; if 'yes' allow incoming calls waiting; by default use network settings
	; if 'no' waiting calls just ignored
	disable=no ; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section

	initstate=start ; specified initial state of device, must be one of 'stop' 'start' 'remote'
	; 'remove' same as 'disable=yes'

	exten=+375441111111 ; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)

	dtmf=inband ; control of incoming DTMF detection, possible values:
	; off - off DTMF tones detection, voice data passed to asterisk unaltered
	; use this value for gateways or if not use DTMF for AVR or inside dialplan
	; inband - do DTMF tones detection
	; relax - like inband but with relaxdtmf option
	; default is 'relax' by compatibility reason

	; dongle required settings
	[dongle0]
	audio=/dev/ttyUSB1 ; tty port for audio connection; no default value
	data=/dev/ttyUSB2 ; tty port for AT commands; no default value

	context=dongle-incoming; контекст
	group=0

	; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345
	; imei and imsi must contain exactly 15 digits !
	; imei/imsi discovery is available on Linux only
	imei=123456789012345 ; выставлены - правильно
	imsi=123456789012345 ; выставлены - правильно - здесь не показываю просто

	; if audio and data set together with imei and/or imsi audio and data has precedence
	; you can use both imei and imsi together in this case exact match by imei and imsi required
	[general]
	static=yes
	writeprotect=no
	clearglobalvars=no

	[globals]
	CONSOLE=Console/dsp



	[dongle-incoming-ussd]
	exten => ussd,1,Noop(Incoming USSD: ${BASE64_DECODE(${USSD_BASE64})})
	exten => ussd,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME}: ${BASE64_DECODE(${USSD_BASE64})}' >> /var/log/asterisk/ussd.txt)
	exten => ussd,n,Hangup()

	[dongle-incoming-sms]
	exten => sms,1,Noop(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
	exten => sms,n,System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}' >> /var/log/asterisk/sms.txt)
	exten => sms,n,Hangup()

	[dongle-incoming]
	include => dongle-incoming-ussd
	include => dongle-incoming-sms

	;exten => +375441111111,1,Hangup()

	exten => +375441111111,1,NoOp()
	same => n,Wait(1)
	;same => n,Answer()
	;same => n,WaitExten(35)


	;exten => _7XXXXXXXXXX,1,NoOp()
	;same => n,Dial(SIP/10)
	;same => n,Dial(SIP/74951234567@zadarma)
	same => n,Dial(SIP/00000@zadarma)

	same => n,Hangup()



	[default]
	exten => 10,1,Dial(Dongle/dongle0/1111111)
	exten => 00000,1,Dial(SIP/00000@zadarma)
	[general]

	limitonpeer=yes
	notifyringing=yes
	notifyhold=yes
	allowsubscribe=yes


	context=default
	allowguest=no ; no guests
	allowoverlap=no ; allow call with additional numbers -> no
	alwaysauthreject=yes ; all errors from client gives to him as auth errors(yes), no - disabling this feature
	; defaultexpiry=360 ; interval mandatory registration
	callevents=yes ; for future. For activating music on hold
	tcpenable=yes ; enabling TCP for SIP
	rtptimeout=60 ; timeout before kick
	language=ru ; language for voice subsystem
	bindport=5060
	bindaddr=0.0.0.0
	;srvlookup=yes ; searching for server address by username <testuser@domain.com>
	tos_sip=cs3
	tos_audio=ef

	match_auth_username=yes

	;video=yes
	;videosupport=yes

	; echocancel=yes
	; echocancelwhenbridged=yes

	disallow=all
	;allow=alaw
	;allow=g729
	;allow=g723
	;allow=ulaw

	canreinvite=no


	pedantic=no

	[zadarma-tpl](!)
	type=friend
	host=sipde.zadarma.com
	fromdomain=sip.zadarma.com
	nat=force_rport,comedia
	;directmedia=nonat
	;nat=yes
	dtmfmode=rfc2833
	insecure=invite
	canreinvite=no
	qualify=yes
	disallow=all
	allow=alaw
	allow=ulaw
	allow=gsm

	[zadarma](zadarma-tpl)
	context=from-zadarma
	username=11111
	secret=xxxxx
	fromuser=11111


	[defaults](!)
	canreinvite=no
	disallow=all
	;allow=g729
	allow=alaw
	allow=ulaw
	;allow=ilbc
	;allow=g726
	;allow=g723
	;allow=g722
	allow=gsm
	nat=force_rport,comedia
	directmedia=nonat

	[peer](!,defaults)
	type=friend
	dtmfmode=rfc2833
	host=dynamic
	qualify=yes
	call-limit=3
	busylevel=1
	rtpkeepalive=5


	[10](peer) ; username
	callerid=username
	defaultuser=username
	secret=00000